Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/

Bug: webrtc:13555, webrtc:13082
Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35771}

modules/audio_coding/neteq/tools/audio_checksum.h

@@ -16,7 +16,6 @@
 
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/message_digest.h"
 #include "rtc_base/string_encode.h"
 #include "rtc_base/system/arch.h"
@@ -31,6 +30,9 @@ class AudioChecksum : public AudioSink {
         checksum_result_(checksum_->Size()),
         finished_(false) {}
 
+  AudioChecksum(const AudioChecksum&) = delete;
+  AudioChecksum& operator=(const AudioChecksum&) = delete;
+
   bool WriteArray(const int16_t* audio, size_t num_samples) override {
     if (finished_)
       return false;
@@ -56,8 +58,6 @@ class AudioChecksum : public AudioSink {
   std::unique_ptr<rtc::MessageDigest> checksum_;
   rtc::Buffer checksum_result_;
   bool finished_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioChecksum);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/audio_loop.h

@@ -15,7 +15,6 @@
 #include <string>
 
 #include "api/array_view.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -29,6 +28,9 @@ class AudioLoop {
 
   virtual ~AudioLoop() {}
 
+  AudioLoop(const AudioLoop&) = delete;
+  AudioLoop& operator=(const AudioLoop&) = delete;
+
   // Initializes the AudioLoop by reading from `file_name`. The loop will be no
   // longer than `max_loop_length_samples`, if the length of the file is
   // greater. Otherwise, the loop length is the same as the file length.
@@ -47,8 +49,6 @@ class AudioLoop {
   size_t loop_length_samples_;
   size_t block_length_samples_;
   std::unique_ptr<int16_t[]> audio_array_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioLoop);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/audio_sink.h

@@ -12,7 +12,6 @@
 #define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
 
 #include "api/audio/audio_frame.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -24,6 +23,9 @@ class AudioSink {
   AudioSink() {}
   virtual ~AudioSink() {}
 
+  AudioSink(const AudioSink&) = delete;
+  AudioSink& operator=(const AudioSink&) = delete;
+
   // Writes `num_samples` from `audio` to the AudioSink. Returns true if
   // successful, otherwise false.
   virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
@@ -34,9 +36,6 @@ class AudioSink {
     return WriteArray(audio_frame.data(), audio_frame.samples_per_channel_ *
                                               audio_frame.num_channels_);
   }
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioSink);
 };
 
 // Forks the output audio to two AudioSink objects.
@@ -45,23 +44,25 @@ class AudioSinkFork : public AudioSink {
   AudioSinkFork(AudioSink* left, AudioSink* right)
       : left_sink_(left), right_sink_(right) {}
 
+  AudioSinkFork(const AudioSinkFork&) = delete;
+  AudioSinkFork& operator=(const AudioSinkFork&) = delete;
+
   bool WriteArray(const int16_t* audio, size_t num_samples) override;
 
  private:
   AudioSink* left_sink_;
   AudioSink* right_sink_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
 };
 
 // An AudioSink implementation that does nothing.
 class VoidAudioSink : public AudioSink {
  public:
   VoidAudioSink() = default;
-  bool WriteArray(const int16_t* audio, size_t num_samples) override;
 
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(VoidAudioSink);
+  VoidAudioSink(const VoidAudioSink&) = delete;
+  VoidAudioSink& operator=(const VoidAudioSink&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override;
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/constant_pcm_packet_source.h

@@ -16,7 +16,6 @@
 #include <string>
 
 #include "modules/audio_coding/neteq/tools/packet_source.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -31,6 +30,9 @@ class ConstantPcmPacketSource : public PacketSource {
                           int sample_rate_hz,
                           int payload_type);
 
+  ConstantPcmPacketSource(const ConstantPcmPacketSource&) = delete;
+  ConstantPcmPacketSource& operator=(const ConstantPcmPacketSource&) = delete;
+
   std::unique_ptr<Packet> NextPacket() override;
 
  private:
@@ -46,8 +48,6 @@ class ConstantPcmPacketSource : public PacketSource {
   uint16_t seq_number_;
   uint32_t timestamp_;
   const uint32_t payload_ssrc_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(ConstantPcmPacketSource);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/input_audio_file.h

@@ -15,8 +15,6 @@
 
 #include <string>
 
-#include "rtc_base/constructor_magic.h"
-
 namespace webrtc {
 namespace test {
 
@@ -27,6 +25,9 @@ class InputAudioFile {
 
   virtual ~InputAudioFile();
 
+  InputAudioFile(const InputAudioFile&) = delete;
+  InputAudioFile& operator=(const InputAudioFile&) = delete;
+
   // Reads `samples` elements from source file to `destination`. Returns true
   // if the read was successful, otherwise false. If the file end is reached,
   // the file is rewound and reading continues from the beginning.
@@ -52,7 +53,6 @@ class InputAudioFile {
  private:
   FILE* fp_;
   const bool loop_at_end_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(InputAudioFile);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/output_audio_file.h

@@ -16,7 +16,6 @@
 #include <string>
 
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -34,6 +33,9 @@ class OutputAudioFile : public AudioSink {
       fclose(out_file_);
   }
 
+  OutputAudioFile(const OutputAudioFile&) = delete;
+  OutputAudioFile& operator=(const OutputAudioFile&) = delete;
+
   bool WriteArray(const int16_t* audio, size_t num_samples) override {
     RTC_DCHECK(out_file_);
     return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
@@ -41,8 +43,6 @@ class OutputAudioFile : public AudioSink {
 
  private:
   FILE* out_file_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/output_wav_file.h

@@ -15,7 +15,6 @@
 
 #include "common_audio/wav_file.h"
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -29,6 +28,9 @@ class OutputWavFile : public AudioSink {
                 int num_channels = 1)
       : wav_writer_(file_name, sample_rate_hz, num_channels) {}
 
+  OutputWavFile(const OutputWavFile&) = delete;
+  OutputWavFile& operator=(const OutputWavFile&) = delete;
+
   bool WriteArray(const int16_t* audio, size_t num_samples) override {
     wav_writer_.WriteSamples(audio, num_samples);
     return true;
@@ -36,8 +38,6 @@ class OutputWavFile : public AudioSink {
 
  private:
   WavWriter wav_writer_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/packet.h

@@ -16,7 +16,6 @@
 #include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/copy_on_write_buffer.h"
 
 namespace webrtc {
@@ -54,6 +53,9 @@ class Packet {
 
   virtual ~Packet();
 
+  Packet(const Packet&) = delete;
+  Packet& operator=(const Packet&) = delete;
+
   // Parses the first bytes of the RTP payload, interpreting them as RED headers
   // according to RFC 2198. The headers will be inserted into `headers`. The
   // caller of the method assumes ownership of the objects in the list, and
@@ -95,8 +97,6 @@ class Packet {
   size_t virtual_payload_length_bytes_ = 0;
   const double time_ms_;     // Used to denote a packet's arrival time.
   const bool valid_header_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Packet);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/packet_source.h

@@ -15,7 +15,6 @@
 #include <memory>
 
 #include "modules/audio_coding/neteq/tools/packet.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -26,6 +25,9 @@ class PacketSource {
   PacketSource();
   virtual ~PacketSource();
 
+  PacketSource(const PacketSource&) = delete;
+  PacketSource& operator=(const PacketSource&) = delete;
+
   // Returns next packet. Returns nullptr if the source is depleted, or if an
   // error occurred.
   virtual std::unique_ptr<Packet> NextPacket() = 0;
@@ -34,9 +36,6 @@ class PacketSource {
 
  protected:
   std::bitset<128> filter_;  // Payload type is 7 bits in the RFC.
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/resample_input_audio_file.h

@@ -15,7 +15,6 @@
 
 #include "common_audio/resampler/include/resampler.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -37,6 +36,9 @@ class ResampleInputAudioFile : public InputAudioFile {
         file_rate_hz_(file_rate_hz),
         output_rate_hz_(output_rate_hz) {}
 
+  ResampleInputAudioFile(const ResampleInputAudioFile&) = delete;
+  ResampleInputAudioFile& operator=(const ResampleInputAudioFile&) = delete;
+
   bool Read(size_t samples, int output_rate_hz, int16_t* destination);
   bool Read(size_t samples, int16_t* destination) override;
   void set_output_rate_hz(int rate_hz);
@@ -45,7 +47,6 @@ class ResampleInputAudioFile : public InputAudioFile {
   const int file_rate_hz_;
   int output_rate_hz_;
   Resampler resampler_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/rtc_event_log_source.h

@@ -19,7 +19,6 @@
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/audio_coding/neteq/tools/packet_source.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -43,6 +42,9 @@ class RtcEventLogSource : public PacketSource {
 
   virtual ~RtcEventLogSource();
 
+  RtcEventLogSource(const RtcEventLogSource&) = delete;
+  RtcEventLogSource& operator=(const RtcEventLogSource&) = delete;
+
   std::unique_ptr<Packet> NextPacket() override;
 
   // Returns the timestamp of the next audio output event, in milliseconds. The
@@ -60,8 +62,6 @@ class RtcEventLogSource : public PacketSource {
   size_t rtp_packet_index_ = 0;
   std::vector<int64_t> audio_outputs_;
   size_t audio_output_index_ = 0;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/rtp_file_source.h

@@ -19,7 +19,6 @@
 #include "absl/types/optional.h"
 #include "modules/audio_coding/neteq/tools/packet_source.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -41,6 +40,9 @@ class RtpFileSource : public PacketSource {
 
   ~RtpFileSource() override;
 
+  RtpFileSource(const RtpFileSource&) = delete;
+  RtpFileSource& operator=(const RtpFileSource&) = delete;
+
   // Registers an RTP header extension and binds it to `id`.
   virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
 
@@ -58,8 +60,6 @@ class RtpFileSource : public PacketSource {
   std::unique_ptr<RtpFileReader> rtp_reader_;
   const absl::optional<uint32_t> ssrc_filter_;
   RtpHeaderExtensionMap rtp_header_extension_map_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
 };
 
 }  // namespace test

modules/audio_coding/neteq/tools/rtp_generator.h

@@ -12,7 +12,6 @@
 #define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
 
 #include "api/rtp_headers.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -34,6 +33,9 @@ class RtpGenerator {
 
   virtual ~RtpGenerator() {}
 
+  RtpGenerator(const RtpGenerator&) = delete;
+  RtpGenerator& operator=(const RtpGenerator&) = delete;
+
   // Writes the next RTP header to `rtp_header`, which will be of type
   // `payload_type`. Returns the send time for this packet (in ms). The value of
   // `payload_length_samples` determines the send time for the next packet.
@@ -50,9 +52,6 @@ class RtpGenerator {
   const uint32_t ssrc_;
   const int samples_per_ms_;
   double drift_factor_;
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
 };
 
 class TimestampJumpRtpGenerator : public RtpGenerator {
@@ -66,6 +65,10 @@ class TimestampJumpRtpGenerator : public RtpGenerator {
         jump_from_timestamp_(jump_from_timestamp),
         jump_to_timestamp_(jump_to_timestamp) {}
 
+  TimestampJumpRtpGenerator(const TimestampJumpRtpGenerator&) = delete;
+  TimestampJumpRtpGenerator& operator=(const TimestampJumpRtpGenerator&) =
+      delete;
+
   uint32_t GetRtpHeader(uint8_t payload_type,
                         size_t payload_length_samples,
                         RTPHeader* rtp_header) override;
@@ -73,7 +76,6 @@ class TimestampJumpRtpGenerator : public RtpGenerator {
  private:
   uint32_t jump_from_timestamp_;
   uint32_t jump_to_timestamp_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
 };
 
 }  // namespace test